[Asterisk-Users] Bridged line appearance

Discussion:

mustardman29

2006-02-18 06:05:52 UTC

So are there any plans for bridged line appearance support in Asterisk? The
new Linksys SPA9000 supports it. A lot of other VoIP systems from Nortel,
Sylantro etc. supposedly support it. Seems to me that Asterisk needs to
get on the bandwagon or be relegated to call centers, specialized voicemail
applications, and phone chat businesses. It's not needed for companies used
to PBX's but something like 75-95% of all companies are small businesses
using key systems with BLA type behaviour not PBX behaviour.

Like it or not, the mass market uses and will continue to use BLA or
whatever they call it in the non VoIP world. I know that without it,
Asterisk is a non-starter for most small businesses looking to replace their
key systems.

I am not a software developer but I remember reading a post by an asterisk
developer stating that implementing it in Asterisk would be difficult but
without it I think mass market appeal of Asterisk will be quite limited
IMHO. There is a famous quote that states, "nothing worth doing is ever
easy".

Aastra just released their v1.3.1 firmware which supposedly supports the
internet draft spec of BLA. The Polycom phones also supposedly support this
spec so the ability IS there on the phone side.

BJ Weschke

2006-02-18 08:43:57 UTC

Permalink

Post by mustardman29
So are there any plans for bridged line appearance support in Asterisk? The
new Linksys SPA9000 supports it. A lot of other VoIP systems from Nortel,
Sylantro etc. supposedly support it. Seems to me that Asterisk needs to
get on the bandwagon or be relegated to call centers, specialized voicemail
applications, and phone chat businesses. It's not needed for companies used
to PBX's but something like 75-95% of all companies are small businesses
using key systems with BLA type behaviour not PBX behaviour.
Like it or not, the mass market uses and will continue to use BLA or
whatever they call it in the non VoIP world. I know that without it,
Asterisk is a non-starter for most small businesses looking to replace their
key systems.
I am not a software developer but I remember reading a post by an asterisk
developer stating that implementing it in Asterisk would be difficult but
without it I think mass market appeal of Asterisk will be quite limited
IMHO. There is a famous quote that states, "nothing worth doing is ever
easy".
Aastra just released their v1.3.1 firmware which supposedly supports the
internet draft spec of BLA. The Polycom phones also supposedly support this
spec so the ability IS there on the phone side.

1) Yes. There are "plans for it".

2) No. It won't be easy as Asterisk is a multi-protocol PBX and
usually when we consider introducing a feature like this, its intent
is for it to function across all of the protocols that Asterisk
supports, VoIP or not. Everyone else you've mentioned needs only worry
about their own device supporting a standard or their own system only
supporting devices that they manufacture to support the feature. That
makes things somewhat easier for implementation and Asterisk has no
such luxury given it's completely open nature which most of us see as
an advantage.

3) The other solutions you've mentioned above all have salaried
engineering staffs whose job it is to implement features as decided by
product management folks also employed by that company who are driven
by the comments and feedback of users such as yourself who fork over
large sums of money compared to what you pay for your Asterisk to have
such solutions. Had you sent such an email to one of these companies
at the time you did on a Friday night in the states, my bet is on the
fact that it wouldn't have even solicited an initial response from a
product management resource until Monday morning.

4) The SPA-9000 is devoid of features like, Voicemail, which Asterisk
already has. If a system without BLA is a "non-starter" for you and
these small business you have cited, why not consider a combined
solution where Asterisk provides features (call queues/ACD, voicemail,
etc) that the SPA-9000 does not have and then you use the SPA-9000 for
what it is good for (an IP key system - which is not what Asterisk
is)? Asterisk can be whatever and play whatever part you want it to
play in your solution. It doesn't have to be the entire solution.
Because of its open nature, it usually integrates and interoperates
well with many existing products/solutions. The SPA-9000 is no
exception.

5) There are thousands of small businesses already, my own being one
of them, that would disagree that Asterisk is a "non starter" for
them. Asterisk is what you make of it, and for us, it's a criticial
communications tool for our business.

These things being said, what was your original intent for writing
such an email? Is there something you'd like to contribute to help get
this feature implemented? You don't need to be a developer to
contribute. There's testing, documentation, bounties to be set for
features one "must have", and all sorts of other areas that could use
the assistance of folks like yourself that aren't software developers.

Thanks for your initial feedback and we look forward to your
continued contributions to the Asterisk community.

--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/

mustardman29

2006-02-18 17:59:18 UTC

Permalink

Post by BJ Weschke
1) Yes. There are "plans for it".

GREAT! What is the current status and expected timeline?

Post by BJ Weschke
2) No. It won't be easy as Asterisk is a multi-protocol PBX
and usually when we consider introducing a feature like this,
its intent is for it to function across all of the protocols
that Asterisk supports, VoIP or not. Everyone else you've
mentioned needs only worry about their own device supporting
a standard or their own system only supporting devices that
they manufacture to support the feature. That makes things
somewhat easier for implementation and Asterisk has no such
luxury given it's completely open nature which most of us see
as an advantage.

Thanks for explaining the details of why it will be difficult

Post by BJ Weschke
3) The other solutions you've mentioned above all have
salaried engineering staffs whose job it is to implement
features as decided by product management folks also employed
by that company who are driven by the comments and feedback
of users such as yourself who fork over large sums of money
compared to what you pay for your Asterisk to have such
solutions. Had you sent such an email to one of these
companies at the time you did on a Friday night in the
states, my bet is on the fact that it wouldn't have even
solicited an initial response from a product management
resource until Monday morning.

Ummm.....ok. Asterisk=open source community. That just goes without
saying. Other than that I don't know what your point is. So there are no
salaried software engineers at Digium working on Asterisk?

Post by BJ Weschke
4) The SPA-9000 is devoid of features like, Voicemail, which
Asterisk already has. If a system without BLA is a
"non-starter" for you and these small business you have
cited, why not consider a combined solution where Asterisk
provides features (call queues/ACD, voicemail,
etc) that the SPA-9000 does not have and then you use the
SPA-9000 for what it is good for (an IP key system - which is
not what Asterisk is)? Asterisk can be whatever and play
whatever part you want it to play in your solution. It
doesn't have to be the entire solution.
Because of its open nature, it usually integrates and
interoperates well with many existing products/solutions. The
SPA-9000 is no exception.

Thanks for pointing out the differences. Yes, I have thought about creating
a Frankenstein system which takes advantage of the strengths of both the
SPA-9000 and Asterisk. Perhaps using Asterisk as a POT's gateway and
voicemail server. The cost starts to creep up though. This is a concept I
have been mulling over for awhile now. It remains to be seen what the best
direction is. When in doubt the best strategy is KISS. The simplest,
cheapest, and presumably most robust solution is to have everything in one
box.

Post by BJ Weschke
5) There are thousands of small businesses already, my own
being one of them, that would disagree that Asterisk is a
"non starter" for them. Asterisk is what you make of it, and
for us, it's a criticial communications tool for our business.

At the end of the day it is what the user thinks, not the Linux people. For
you, me and most others on this board I think we can all agree that Asterisk
works just fine for us. For some companies used to PBX like functionality
it will probably work just fine as well which I have already pointed out.
For many many other companies used to key system like functionality it is a
non-starter mostly because of the lack of BLA IMHO. If you don't believe me
that it is a VERY important feature then ask yourself why a LOT of IP phones
and VoIP systems support it or are starting to support it. If Asterisk
wants to be a main stream phone system then I feel it should support it.
Has nothing to do with open source vs proprietary. Just giving my opinion
based on user feedback.

Post by BJ Weschke
These things being said, what was your original intent for
writing such an email? Is there something you'd like to
contribute to help get this feature implemented? You don't
need to be a developer to contribute. There's testing,
documentation, bounties to be set for features one "must
have", and all sorts of other areas that could use the
assistance of folks like yourself that aren't software developers.

Sure, what is the development schedule? I get your point. No need to beat
me over the head with it. I read these sorts of comments about how it's
"your fault for not being a software coder" and "if you don't like it too
bad, it's your fault for not getting more involved" and frankly I am sick of
it. We all know this is open source, we mostly all know the advantages and
disadvantages of it and we would not be here if we didn't want it to work.
Let's just move on. I am sorry for not being able to code. I am sorry I am
not contributing as much as I should. It's my fault this feature is not
getting off the ground. There are you happy? Can we move on now?

Post by BJ Weschke
Thanks for your initial feedback and we look forward to your
continued contributions to the Asterisk community.
--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/

ADEGOKE ARUNA

2006-02-18 19:40:32 UTC

Permalink

Hi all,

I will be glad if I can get response to my need.

What I am trying to do is;

I have a set up where my asterisk box is directly connected to a digitalk IN
platform.

However, between the asterisk and digitalk is a104d sangoma card having e1
with pri. The link between digitalk and my big alcatel switch is e1 with ss7
signalling and this finally lead to my mobile operator

What I am planning to do is to rewrite the caller id so that the calling
number presented to the mobile operator is going to be the number set on the
pri channels

I will be glad, if anyone can just lead me on this

goksie

Jean-Christophe Heger

2006-02-18 19:53:36 UTC

Permalink

You may change the CallerID with SetCallerID function, and the
presentation with CallingPres, before dialing.

In theory, you may place the CallerID you want, but your phone operator
could refuse it if it doesn't belong to you (will show a default
number). You also might have to use CallingPres to tune the presentation
flags. For Swisscom with a ZapHFC BRI card, working values are 0: show,
and 32: hide.

Jean-Christophe

Post by ADEGOKE ARUNA
Hi all,
I will be glad if I can get response to my need.
What I am trying to do is;
I have a set up where my asterisk box is directly connected to a digitalk IN
platform.
However, between the asterisk and digitalk is a104d sangoma card having e1
with pri. The link between digitalk and my big alcatel switch is e1 with ss7
signalling and this finally lead to my mobile operator
What I am planning to do is to rewrite the caller id so that the calling
number presented to the mobile operator is going to be the number set on the
pri channels
I will be glad, if anyone can just lead me on this
goksie
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David Ankers

2006-02-18 20:02:35 UTC

Permalink

Simply amazing.

-----Original Message-----
From: asterisk-users-***@lists.digium.com
[mailto:asterisk-users-***@lists.digium.com] On Behalf Of mustardman29
Sent: Sunday, 19 February 2006 4:59 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Bridged line appearance

Post by BJ Weschke
1) Yes. There are "plans for it".

GREAT! What is the current status and expected timeline?

Thanks for explaining the details of why it will be difficult

Ummm.....ok. Asterisk=open source community. That just goes without
saying. Other than that I don't know what your point is. So there are no
salaried software engineers at Digium working on Asterisk?

Post by BJ Weschke
Thanks for your initial feedback and we look forward to your
continued contributions to the Asterisk community.
--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/

Michael J. Liberatore

2006-02-18 23:23:51 UTC

Permalink

Man, I am all for shared line appearances. I have asterisk systems in
several small businesses and they all cry for it. But there are ways
around it as well, after a week all the bussinesses have gotten used to
asterisk w/o bla. Plus, past 4 lines, its hard to implement cause lots
of phones only have 4 lines. Trust me though arguing on this list wont
get you the feature quicker, I have read tons of e-mails on here and
have seen a pattern :)

Now, I don't code C, but would like the feature for some customers. If
you would be interested in forming a bounty with me, I would be possibly
willing to donate some money to the bounty with you. But if you just
want to complain then good luck getting this implemented quicker.

Mike

-----Original Message-----
From: asterisk-users-***@lists.digium.com
[mailto:asterisk-users-***@lists.digium.com] On Behalf Of
mustardman29
Sent: Saturday, February 18, 2006 12:59 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Bridged line appearance

Post by BJ Weschke
1) Yes. There are "plans for it".

GREAT! What is the current status and expected timeline?

Post by BJ Weschke
2) No. It won't be easy as Asterisk is a multi-protocol PBX and
usually when we consider introducing a feature like this, its intent
is for it to function across all of the protocols that Asterisk
supports, VoIP or not. Everyone else you've mentioned needs only worry
about their own device supporting a standard or their own system only
supporting devices that they manufacture to support the feature. That
makes things somewhat easier for implementation and Asterisk has no
such luxury given it's completely open nature which most of us see as
an advantage.

Thanks for explaining the details of why it will be difficult

Ummm.....ok. Asterisk=open source community. That just goes without
saying. Other than that I don't know what your point is. So there are
no
salaried software engineers at Digium working on Asterisk?

Thanks for pointing out the differences. Yes, I have thought about
creating
a Frankenstein system which takes advantage of the strengths of both the
SPA-9000 and Asterisk. Perhaps using Asterisk as a POT's gateway and
voicemail server. The cost starts to creep up though. This is a
concept I
have been mulling over for awhile now. It remains to be seen what the
best
direction is. When in doubt the best strategy is KISS. The simplest,
cheapest, and presumably most robust solution is to have everything in
one
box.

At the end of the day it is what the user thinks, not the Linux people.
For
you, me and most others on this board I think we can all agree that
Asterisk
works just fine for us. For some companies used to PBX like
functionality
it will probably work just fine as well which I have already pointed
out.
For many many other companies used to key system like functionality it
is a
non-starter mostly because of the lack of BLA IMHO. If you don't believe
me
that it is a VERY important feature then ask yourself why a LOT of IP
phones
and VoIP systems support it or are starting to support it. If Asterisk
wants to be a main stream phone system then I feel it should support it.
Has nothing to do with open source vs proprietary. Just giving my
opinion
based on user feedback.

Sure, what is the development schedule? I get your point. No need to
beat
me over the head with it. I read these sorts of comments about how it's
"your fault for not being a software coder" and "if you don't like it
too
bad, it's your fault for not getting more involved" and frankly I am
sick of
it. We all know this is open source, we mostly all know the advantages
and
disadvantages of it and we would not be here if we didn't want it to
work.
Let's just move on. I am sorry for not being able to code. I am sorry
I am
not contributing as much as I should. It's my fault this feature is not
getting off the ground. There are you happy? Can we move on now?

Post by BJ Weschke
Thanks for your initial feedback and we look forward to your
continued contributions to the Asterisk community.
--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/

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John Novack

2006-02-18 23:48:46 UTC

Permalink

Many very low cost hybrid key/pbx systems for the small business SOHO
market have 12 or more programmable buttons, so regardless of what is
done with Asterisk, until the IP phone manufacturers take off their
blinders and manufacture competing equipment, this market will be out of
reach. These same systems now have voice mail systems with capabilities
and features that make Comedian Mail the correct name. Asterisk isn't
alone regarding these shortfalls, of course. IP phone system designers
have failed to understand the small business market for several years.

Defensive responses with lengthy explanations why it can't be done, or
why it hasn't been done and will be difficult miss the point. Either
Asterisk needs to change to move into this market, or another product will

JMO

John Novack

Clint Sharp

2006-02-19 01:20:20 UTC

Permalink

I'm having a very hard time justifying trying to sell this to the SOHO
market on price or parity with key systems. I've installed key systems and
large scale PBXs, and while working around the SLA problem isn't that hard,
the price point for a key system is very hard to compete with. I've never
understood why people would want to use an SLA system, honestly, as it's a
really poor model. I hate sitting in offices with constant paging "Call for
blah, line 1". The PBX model to me is much more preferable, and working
around it is simply a training problem.

The problem with asterisk isn't the lack of SLA, it's the price point. It's
going to be very hard for IP phone vendors to compete on price at this
point, and so far the quality issues in low-priced hardware to me means I
can't really sell this to anyone who's not willing to pay $200-$300 a phone
(retail).

Not that it's impossible, it's a different sales strategy. Perhaps people
who are wanting to sell this to the SOHO market should attempt to change the
game, as PBX like functionality doesn't exist in the SOHO market because it
hasn't been affordable previously. Asterisk systems are pretty cheap in
terms of the features they offer, such that the sales pitch really depends
on cost for features and maintenance and infrastructure savings rather than
overall cost.

Admittedly though, the voicemail system's navigation issues are a big
problem.

Clint

Post by John Novack
Many very low cost hybrid key/pbx systems for the small business SOHO
market have 12 or more programmable buttons, so regardless of what is
done with Asterisk, until the IP phone manufacturers take off their
blinders and manufacture competing equipment, this market will be out of
reach. These same systems now have voice mail systems with capabilities
and features that make Comedian Mail the correct name. Asterisk isn't
alone regarding these shortfalls, of course. IP phone system designers
have failed to understand the small business market for several years.
Defensive responses with lengthy explanations why it can't be done, or
why it hasn't been done and will be difficult miss the point. Either
Asterisk needs to change to move into this market, or another product will
JMO
John Novack
_______________________________________________
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Asterisk-Users mailing list
http://lists.digium.com/mailman/listinfo/asterisk-users

Adam Robins

2006-02-19 01:24:19 UTC

Permalink

After many days of playing with the new jitterbuffer and trunking options for IAX2, I have finally received almost acceptable quality. I am receiving 5-8 complaints a day of calls "breaking up" from both the customer and agent sides. What I have discovered is that in most of these cases, the new jitterbuffer performed a resync during the call. Currently, I have the resyncthreshold, and all other jb parameters at their default levels The traffic is running over a fairly high latency WAN connection between Canada and Atlanta (IAX2, ILBC). Idle ping times run about 85ms.

Below are the resync messages for this past Friday. Knowing that I have a slow connection, should I set the resync at a much higher level? I appreciate any assistance you may provide.

Thanks,
Adam

Feb 17 09:07:41 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -34, this delay 1651, threshold 1488, new offset -1651
Feb 17 09:07:42 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -120, this delay -1684, threshold 1000, new offset 33
Feb 17 10:21:04 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 176, this delay 1835, threshold 1126, new offset -1835
Feb 17 10:21:04 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 32, this delay 1673, threshold 1062, new offset -1673
Feb 17 10:21:04 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -150, this delay -1663, threshold 1300, new offset -172
Feb 17 10:21:04 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -150, this delay -1635, threshold 1300, new offset -38
Feb 17 10:21:48 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -22, this delay 2335, threshold 1054, new offset -2373
Feb 17 10:21:48 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 11, this delay 2363, threshold 1082, new offset -2535
Feb 17 10:21:48 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -71, this delay 2249, threshold 1054, new offset -2249
Feb 17 10:21:48 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -180, this delay -2359, threshold 1360, new offset -14
Feb 17 10:21:48 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -150, this delay -2354, threshold 1300, new offset -181
Feb 17 10:21:48 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -120, this delay -2297, threshold 1240, new offset 48
Feb 17 10:34:28 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 109, this delay 1556, threshold 1136, new offset -1556
Feb 17 10:34:28 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -30, this delay -1439, threshold 1000, new offset -117
Feb 17 10:34:32 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -7, this delay 1608, threshold 1048, new offset -1725
Feb 17 10:34:32 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -29, this delay -1616, threshold 1058, new offset -109
Feb 17 10:45:08 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 21, this delay 1751, threshold 1620, new offset -1751
Feb 17 10:45:08 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -7, this delay 1724, threshold 1686, new offset -1724
Feb 17 10:45:08 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -60, this delay -1716, threshold 1000, new offset -8
Feb 17 10:45:08 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -119, this delay -1757, threshold 1000, new offset 6
Feb 17 11:28:45 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 75, this delay 1421, threshold 1326, new offset -1421
Feb 17 11:28:45 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 274, this delay 1595, threshold 1282, new offset -1595
Feb 17 11:29:03 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -1311, this delay 820, threshold 1824, new offset -2415
Feb 17 11:29:03 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -1349, this delay 761, threshold 1752, new offset -2182
Feb 17 11:29:03 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -299, this delay -2127, threshold 1598, new offset -288
Feb 17 11:29:03 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -270, this delay -2106, threshold 1540, new offset -76
Feb 17 11:46:15 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 98, this delay 1878, threshold 1206, new offset -1878
Feb 17 11:46:15 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 44, this delay 1799, threshold 1150, new offset -1799
Feb 17 11:46:15 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 28, this delay 1781, threshold 1146, new offset -1781
Feb 17 11:46:15 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -150, this delay -1753, threshold 1000, new offset -46
Feb 17 11:46:15 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -150, this delay -1765, threshold 1000, new offset -16
Feb 17 11:46:15 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -149, this delay -1747, threshold 1298, new offset -131
Feb 17 11:54:36 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -44, this delay 1136, threshold 1064, new offset -1152
Feb 17 11:54:36 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 1, this delay 1155, threshold 1080, new offset -1155
Feb 17 11:54:36 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 405, this delay 1547, threshold 1080, new offset -1547
Feb 17 11:54:36 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -26, this delay 1115, threshold 1054, new offset -1115
Feb 17 11:54:36 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -60, this delay -1133, threshold 1000, new offset -414
Feb 17 11:54:48 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 3, this delay 1144, threshold 1048, new offset -1558

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yusuf

2006-02-20 15:27:05 UTC

Permalink

Adam Robins wrote:
Hi Adam

I am interested to know why you are using ilbc, n why not g729 ot g723
or speex. What is the size of the WAN connection. How many calls are
you running over this link. I just need to see how others are fairing
with IAX2 over WAN links, as I am the final stages of testing on my side

thanks,
yusuf

Peter Fern

2006-02-21 12:58:51 UTC

Permalink

I had exactly the same experience running IAX2, but also experienced
half-duplex calls on top of that (though I think that's a different but
with IAX handoff), and in the end dropped it completely for SIP.

We run g729 over dedicated fibre, and the resyncs were occurring all
over the place with quite ludicrous values logged for delay. I tried
tweaking the jitterbuf, turning it off completely, and reverting to the
old jitterbuffer implementation. none of which made any difference. I
also tried with and without trunking enabled.

SIP is running much more acceptably now.

Post by Adam Robins
After many days of playing with the new jitterbuffer and trunking options for IAX2, I have finally received almost acceptable quality. I am receiving 5-8 complaints a day of calls "breaking up" from both the customer and agent sides. What I have discovered is that in most of these cases, the new jitterbuffer performed a resync during the call. Currently, I have the resyncthreshold, and all other jb parameters at their default levels The traffic is running over a fairly high latency WAN connection between Canada and Atlanta (IAX2, ILBC). Idle ping times run about 85ms.
Below are the resync messages for this past Friday. Knowing that I have a slow connection, should I set the resync at a much higher level? I appreciate any assistance you may provide.
Thanks,
Adam
Feb 17 09:07:41 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -34, this delay 1651, threshold 1488, new offset -1651
Feb 17 09:07:42 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -120, this delay -1684, threshold 1000, new offset 33
Feb 17 10:21:04 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 176, this delay 1835, threshold 1126, new offset -1835
Feb 17 10:21:04 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 32, this delay 1673, threshold 1062, new offset -1673
Feb 17 10:21:04 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -150, this delay -1663, threshold 1300, new offset -172
Feb 17 10:21:04 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -150, this delay -1635, threshold 1300, new offset -38
Feb 17 10:21:48 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -22, this delay 2335, threshold 1054, new offset -2373
Feb 17 10:21:48 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 11, this delay 2363, threshold 1082, new offset -2535
Feb 17 10:21:48 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -71, this delay 2249, threshold 1054, new offset -2249
Feb 17 10:21:48 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -180, this delay -2359, threshold 1360, new offset -14
Feb 17 10:21:48 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -150, this delay -2354, threshold 1300, new offset -181
Feb 17 10:21:48 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -120, this delay -2297, threshold 1240, new offset 48
Feb 17 10:34:28 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 109, this delay 1556, threshold 1136, new offset -1556
Feb 17 10:34:28 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -30, this delay -1439, threshold 1000, new offset -117
Feb 17 10:34:32 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -7, this delay 1608, threshold 1048, new offset -1725
Feb 17 10:34:32 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -29, this delay -1616, threshold 1058, new offset -109
Feb 17 10:45:08 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 21, this delay 1751, threshold 1620, new offset -1751
Feb 17 10:45:08 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -7, this delay 1724, threshold 1686, new offset -1724
Feb 17 10:45:08 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -60, this delay -1716, threshold 1000, new offset -8
Feb 17 10:45:08 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -119, this delay -1757, threshold 1000, new offset 6
Feb 17 11:28:45 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 75, this delay 1421, threshold 1326, new offset -1421
Feb 17 11:28:45 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 274, this delay 1595, threshold 1282, new offset -1595
Feb 17 11:29:03 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -1311, this delay 820, threshold 1824, new offset -2415
Feb 17 11:29:03 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -1349, this delay 761, threshold 1752, new offset -2182
Feb 17 11:29:03 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -299, this delay -2127, threshold 1598, new offset -288
Feb 17 11:29:03 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -270, this delay -2106, threshold 1540, new offset -76
Feb 17 11:46:15 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 98, this delay 1878, threshold 1206, new offset -1878
Feb 17 11:46:15 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 44, this delay 1799, threshold 1150, new offset -1799
Feb 17 11:46:15 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 28, this delay 1781, threshold 1146, new offset -1781
Feb 17 11:46:15 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -150, this delay -1753, threshold 1000, new offset -46
Feb 17 11:46:15 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -150, this delay -1765, threshold 1000, new offset -16
Feb 17 11:46:15 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -149, this delay -1747, threshold 1298, new offset -131
Feb 17 11:54:36 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -44, this delay 1136, threshold 1064, new offset -1152
Feb 17 11:54:36 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 1, this delay 1155, threshold 1080, new offset -1155
Feb 17 11:54:36 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 405, this delay 1547, threshold 1080, new offset -1547
Feb 17 11:54:36 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -26, this delay 1115, threshold 1054, new offset -1115
Feb 17 11:54:36 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -60, this delay -1133, threshold 1000, new offset -414
Feb 17 11:54:48 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 3, this delay 1144, threshold 1048, new offset -1558
The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any.
------------------------------------------------------------------------
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Zach A

2006-02-19 01:19:37 UTC

Permalink

Hi all,

I've tried everything over past month but no success. When somebody
calls in from a PSTN line or cell phone to my asterisk box, which is a
connected to a SIP provider, MoH doesn't work good, it is very choppy.
I've tried native format, silence suppression, ulaw and gsm music files,
changing RTP ports but nothing helped. It plays ok only for incoming SIP
calls from the same provider. Any guess why it doesn't work for other
incoming calls and is always choppy?

Thanks,

Zach A.

mustardman29

2006-02-19 02:26:03 UTC

Permalink

You make some good points Clint,

I honestly don't think that trying to force feed this to the customer as it
is is the way to go. Key systems have been used for many many years and the
market has decided that they are what people want in the lower end. I have
sat in small offices and witnessed the elegant simplicity of a key system.
It's all 1 single button press to do ANYTHING. The button label and the
light beside it tells you everything you need to know. It works! No multi
button sequences or *xx key presses to know. People on this forum might not
have a problem with more complexity in exchange for more flexibility etc.
but I don't think the people on this forum are anything like an average
user.

Perhaps Asterisk will never be appropriate for they low end Key market or
the Key/PBX hybrid market. I don't know. There are IP phones around with
plenty of buttons to do the job. The Aastra9133i has something like 9
programmable buttons in addition to 3 incoming line buttons which is plenty
for most small businesses. Their latest firmware now fully supports BLF and
apparently SLA.

Did I come across as complaining? Just trying to make a case for what I see
as a highly desireable feature. I do take exception to anyone trying paint
a picture of me being an ungrateful open source software user looking for a
free ride. Digium has made PLENTY of money off of me. If I could pay
another $300 to get the features I want I would. It's not all about saving
a few bucks! If I wanted to do that I would buy Bizfon's.

Post by David Ankers
-----Original Message-----
Sent: Saturday, February 18, 2006 5:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Bridged line appearance
I'm having a very hard time justifying trying to sell this to the SOHO
market on price or parity with key systems. I've installed key
systems and large scale PBXs, and while working around the SLA problem
isn't that hard, the price point for a key system is very hard to
compete with. I've never understood why people would want to use an
SLA system, honestly, as it's a really poor model. I hate sitting in
offices with constant paging "Call for blah, line 1". The PBX model
to me is much more preferable, and working around it is simply a
training problem.
The problem with asterisk isn't the lack of SLA, it's the price point.
It's going to be very hard for IP phone vendors to compete on price at
this point, and so far the quality issues in low-priced hardware to me
means I can't really sell this to anyone who's not willing to pay
$200-$300 a phone (retail).
Not that it's impossible, it's a different sales strategy.
Perhaps people who are wanting to sell this to the SOHO market should
attempt to change the game, as PBX like functionality doesn't exist in
the SOHO market because it hasn't been affordable previously.
Asterisk systems are pretty cheap in terms of the features they offer,
such that the sales pitch really depends on cost for features and
maintenance and infrastructure savings rather than overall cost.
Admittedly though, the voicemail system's navigation issues are a big
problem.
Clint
Many very low cost hybrid key/pbx systems for the small business

SOHO

Post by David Ankers
market have 12 or more programmable buttons, so regardless of what

Post by David Ankers
done with Asterisk, until the IP phone manufacturers take off their
blinders and manufacture competing equipment, this market will be

out

Post by David Ankers
of
reach. These same systems now have voice mail systems with
capabilities
and features that make Comedian Mail the correct name.
Asterisk isn't
alone regarding these shortfalls, of course. IP phone system
designers
have failed to understand the small business market for several
years.
Defensive responses with lengthy explanations why it can't be done,
or
why it hasn't been done and will be difficult miss the point. Either
Asterisk needs to change to move into this market, or another

product

Post by David Ankers
will
JMO
John Novack
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Adam Robins

2006-02-20 18:40:56 UTC

Permalink

I was using G729 with Asterisk 1.07. With the new ability to do packet
loss correction with ILBC, I felt I'd give it a try. The new PLC does
not work with G729. I don't use Speex because my softphone does not
support it.

This is a 1.5mb IP-VPN connection with prioritized QOS for port 4569
(IAX2). I've never really stressed the bandwidth. Typically, only
10-20 concurrent calls.

-----Original Message-----
From: asterisk-users-***@lists.digium.com
[mailto:asterisk-users-***@lists.digium.com] On Behalf Of yusuf
Sent: Monday, February 20, 2006 10:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer
Tuning

Adam Robins wrote:
Hi Adam

Post by Adam Robins
After many days of playing with the new jitterbuffer and trunking

Adam Robins

2006-02-20 21:43:14 UTC

Permalink

I have now set the "resyncthreshold" to -1, to turn it off. I have also
set the "maxjitterbuffer" to 2000.

I still received 10 complaints of choppy calls today on Asterisk 1.2.4
versus only 1 complaint on Asterisk 1.07.

-----Original Message-----
From: asterisk-users-***@lists.digium.com
[mailto:asterisk-users-***@lists.digium.com] On Behalf Of yusuf
Sent: Monday, February 20, 2006 10:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer
Tuning

Adam Robins wrote:
Hi Adam

Post by Adam Robins
After many days of playing with the new jitterbuffer and trunking

Jesus E Zepeda

2006-02-20 22:02:39 UTC

Permalink

In my case I don't have a T1 or even a fractional T1, but cable and have
noticed that choppy calls can be reduced by adding tos settings. Like:

Tos=lowdelay|throughput|reliability

Regards,
Jesus

-----Original Message-----
From: Adam Robins [mailto:***@PharmaCentra.com]
Sent: Monday, February 20, 2006 14:43
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer
Tuning

I have now set the "resyncthreshold" to -1, to turn it off. I have also
set the "maxjitterbuffer" to 2000.

I still received 10 complaints of choppy calls today on Asterisk 1.2.4
versus only 1 complaint on Asterisk 1.07.

-----Original Message-----
From: asterisk-users-***@lists.digium.com
[mailto:asterisk-users-***@lists.digium.com] On Behalf Of yusuf
Sent: Monday, February 20, 2006 10:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer
Tuning

Adam Robins wrote:
Hi Adam

Post by Adam Robins
After many days of playing with the new jitterbuffer and trunking

Adam Robins

2006-02-20 23:50:32 UTC

Permalink

Thanks, but we already have the TOS bits set to 0xB8, which matches the QoS settings in our switches and routers.

This is definitely something that changed in the 1.07 to 1.24 upgrade. We have a pair of identical 1.07 servers connected via the same network pipe that do not exhibit these issues.

I might try recompiling with the old jitterbuffer to see if it makes a difference.

________________________________

From: asterisk-users-***@lists.digium.com on behalf of Jesus E Zepeda
Sent: Mon 2/20/2006 5:02 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning

In my case I don't have a T1 or even a fractional T1, but cable and have
noticed that choppy calls can be reduced by adding tos settings. Like:

Tos=lowdelay|throughput|reliability

Regards,
Jesus

-----Original Message-----
From: Adam Robins [mailto:***@PharmaCentra.com]
Sent: Monday, February 20, 2006 14:43
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer
Tuning

I have now set the "resyncthreshold" to -1, to turn it off. I have also
set the "maxjitterbuffer" to 2000.

I still received 10 complaints of choppy calls today on Asterisk 1.2.4
versus only 1 complaint on Asterisk 1.07.

-----Original Message-----
From: asterisk-users-***@lists.digium.com
[mailto:asterisk-users-***@lists.digium.com] On Behalf Of yusuf
Sent: Monday, February 20, 2006 10:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer
Tuning

Adam Robins wrote:
Hi Adam

Post by Adam Robins
After many days of playing with the new jitterbuffer and trunking

Mark Willis

2006-02-21 01:01:58 UTC

Permalink

Post by Adam Robins
This is definitely something that changed in the 1.07 to 1.24
upgrade. We have a pair of identical 1.07 servers connected via the
same network pipe that do not exhibit these issues.
I might try recompiling with the old jitterbuffer to see if it makes a difference.

If you are running trunked IAX, try turning off the jitterbuffer entirely.

Mark

Simone Cittadini

2006-02-23 09:36:35 UTC

Permalink

Post by Adam Robins
Thanks, but we already have the TOS bits set to 0xB8, which matches
the QoS settings in our switches and routers.
This is definitely something that changed in the 1.07 to 1.24
upgrade. We have a pair of identical 1.07 servers connected via the
same network pipe that do not exhibit these issues.
I might try recompiling with the old jitterbuffer to see if it makes a difference.
------------------------------------------------------------------------

Michael J. Liberatore

2006-02-21 00:54:32 UTC

Permalink

so you think this problem is asterisk and not a internet problem? My
customers also complain alot about IAX2 connection to teliax which
seemed to work better in older * versions. I have tried everything with
no success, i switched to sip and its alot better but not perfect...

________________________________

From: asterisk-users-***@lists.digium.com
[mailto:asterisk-users-***@lists.digium.com] On Behalf Of Adam
Robins
Sent: Monday, February 20, 2006 6:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer
Tuning

Thanks, but we already have the TOS bits set to 0xB8, which matches the
QoS settings in our switches and routers.

This is definitely something that changed in the 1.07 to 1.24 upgrade.
We have a pair of identical 1.07 servers connected via the same network
pipe that do not exhibit these issues.

I might try recompiling with the old jitterbuffer to see if it makes a
difference.

________________________________

From: asterisk-users-***@lists.digium.com on behalf of Jesus E
Zepeda
Sent: Mon 2/20/2006 5:02 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer
Tuning

In my case I don't have a T1 or even a fractional T1, but cable and have
noticed that choppy calls can be reduced by adding tos settings. Like:

Tos=lowdelay|throughput|reliability

Regards,
Jesus

-----Original Message-----
From: Adam Robins [mailto:***@PharmaCentra.com]
Sent: Monday, February 20, 2006 14:43
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer
Tuning

I have now set the "resyncthreshold" to -1, to turn it off. I have also
set the "maxjitterbuffer" to 2000.

I still received 10 complaints of choppy calls today on Asterisk 1.2.4
versus only 1 complaint on Asterisk 1.07.

-----Original Message-----
From: asterisk-users-***@lists.digium.com
[mailto:asterisk-users-***@lists.digium.com] On Behalf Of yusuf
Sent: Monday, February 20, 2006 10:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer
Tuning

Adam Robins wrote:
Hi Adam

Post by Adam Robins
After many days of playing with the new jitterbuffer and trunking

options for IAX2, I have finally received almost acceptable quality. I
am receiving 5-8 complaints a day of calls "breaking up" from both the
customer and agent sides. What I have discovered is that in most of
these cases, the new jitterbuffer performed a resync during the call.
Currently, I have the resyncthreshold, and all other jb parameters at
their default levels The traffic is running over a fairly high latency
WAN connection between Canada and Atlanta (IAX2, ILBC). Idle ping times
run about 85ms.
I am interested to know why you are using ilbc, n why not g729 ot g723
or speex. What is the size of the WAN connection. How many calls are
you running over this link. I just need to see how others are fairing
with IAX2 over WAN links, as I am the final stages of testing on my side

thanks,
yusuf
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Adam Robins

2006-02-21 12:33:11 UTC

Permalink

I am not running trunked IAX.

-----Original Message-----
From: asterisk-users-***@lists.digium.com
[mailto:asterisk-users-***@lists.digium.com] On Behalf Of Mark
Willis
Sent: Monday, February 20, 2006 8:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer
Tuning

Post by Adam Robins
This is definitely something that changed in the 1.07 to 1.24 upgrade.
We have a pair of identical 1.07 servers connected via the same
network pipe that do not exhibit these issues.
I might try recompiling with the old jitterbuffer to see if it makes a
difference.

If you are running trunked IAX, try turning off the jitterbuffer
entirely.

Mark

_______________________________________________
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Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
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The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any.

Adam Robins

2006-02-21 12:34:30 UTC

Permalink

This is not going over the Internet. It is going over an MPLS IP-VPN.

________________________________

From: asterisk-users-***@lists.digium.com
[mailto:asterisk-users-***@lists.digium.com] On Behalf Of Michael J.
Liberatore
Sent: Monday, February 20, 2006 7:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer
Tuning

so you think this problem is asterisk and not a internet problem? My
customers also complain alot about IAX2 connection to teliax which
seemed to work better in older * versions. I have tried everything with
no success, i switched to sip and its alot better but not perfect...

________________________________

From: asterisk-users-***@lists.digium.com
[mailto:asterisk-users-***@lists.digium.com] On Behalf Of Adam
Robins
Sent: Monday, February 20, 2006 6:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer
Tuning

Thanks, but we already have the TOS bits set to 0xB8, which matches the
QoS settings in our switches and routers.

This is definitely something that changed in the 1.07 to 1.24 upgrade.
We have a pair of identical 1.07 servers connected via the same network
pipe that do not exhibit these issues.

I might try recompiling with the old jitterbuffer to see if it makes a
difference.

________________________________

From: asterisk-users-***@lists.digium.com on behalf of Jesus E
Zepeda
Sent: Mon 2/20/2006 5:02 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer
Tuning

In my case I don't have a T1 or even a fractional T1, but cable and have
noticed that choppy calls can be reduced by adding tos settings. Like:

Tos=lowdelay|throughput|reliability

Regards,
Jesus

-----Original Message-----
From: Adam Robins [mailto:***@PharmaCentra.com]
Sent: Monday, February 20, 2006 14:43
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer
Tuning

I have now set the "resyncthreshold" to -1, to turn it off. I have also
set the "maxjitterbuffer" to 2000.

I still received 10 complaints of choppy calls today on Asterisk 1.2.4
versus only 1 complaint on Asterisk 1.07.

-----Original Message-----
From: asterisk-users-***@lists.digium.com
[mailto:asterisk-users-***@lists.digium.com] On Behalf Of yusuf
Sent: Monday, February 20, 2006 10:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer
Tuning

Adam Robins wrote:
Hi Adam

Post by Adam Robins
After many days of playing with the new jitterbuffer and trunking

options for IAX2, I have finally received almost acceptable quality. I
am receiving 5-8 complaints a day of calls "breaking up" from both the
customer and agent sides. What I have discovered is that in most of
these cases, the new jitterbuffer performed a resync during the call.
Currently, I have the resyncthreshold, and all other jb parameters at
their default levels The traffic is running over a fairly high latency
WAN connection between Canada and Atlanta (IAX2, ILBC). Idle ping times
run about 85ms.
I am interested to know why you are using ilbc, n why not g729 ot g723
or speex. What is the size of the WAN connection. How many calls are
you running over this link. I just need to see how others are fairing
with IAX2 over WAN links, as I am the final stages of testing on my side

thanks,
yusuf
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Adam Robins

2006-02-21 13:18:37 UTC

Permalink

Thank you for validating that I am not going mad!

I made some additional tweaks for today. We'll see how it goes. If not
well, then I'll try SIP for tomorrow.

Thanks,
Adam

-----Original Message-----
From: asterisk-users-***@lists.digium.com
[mailto:asterisk-users-***@lists.digium.com] On Behalf Of Peter Fern
Sent: Tuesday, February 21, 2006 7:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer
Tuning

I had exactly the same experience running IAX2, but also experienced
half-duplex calls on top of that (though I think that's a different but
with IAX handoff), and in the end dropped it completely for SIP.

We run g729 over dedicated fibre, and the resyncs were occurring all
over the place with quite ludicrous values logged for delay. I tried
tweaking the jitterbuf, turning it off completely, and reverting to the
old jitterbuffer implementation. none of which made any difference. I
also tried with and without trunking enabled.

SIP is running much more acceptably now.

Post by Adam Robins
After many days of playing with the new jitterbuffer and trunking

Post by Adam Robins
Below are the resync messages for this past Friday. Knowing that I

have a slow connection, should I set the resync at a much higher level?
I appreciate any assistance you may provide.

Post by Adam Robins
Thanks,
Adam
Feb 17 09:07:41 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay
-34, this delay 1651, threshold 1488, new offset -1651 Feb 17 09:07:42
WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -120, this
delay -1684, threshold 1000, new offset 33 Feb 17 10:21:04
WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 176, this delay
1835, threshold 1126, new offset -1835 Feb 17 10:21:04 WARNING[1078]
chan_iax2.c: Resyncing the jb. last_delay 32, this delay 1673,
threshold 1062, new offset -1673 Feb 17 10:21:04 WARNING[1078]
chan_iax2.c: Resyncing the jb. last_delay -150, this delay -1663,
threshold 1300, new offset -172 Feb 17 10:21:04 WARNING[1078]
chan_iax2.c: Resyncing the jb. last_delay -150, this delay -1635,
threshold 1300, new offset -38 Feb 17 10:21:48 WARNING[1078]
chan_iax2.c: Resyncing the jb. last_delay -22, this delay 2335,
threshold 1054, new offset -2373 Feb 17 10:21:48 WARNING[1078]
chan_iax2.c: Resyncing the jb. last_delay 11, this delay 2363,
threshold 1082, new offset -2535 Feb 17 10:21:48 WARNING[1078]
chan_iax2.c: Resyncing the jb. last_delay -71, this delay 2249,
threshold 1054, new offset -2249 Feb 17 10:21:48 WARNING[1078]
chan_iax2.c: Resyncing the jb. last_delay -180, this delay -2359,
threshold 1360, new offset -14 Feb 17 10:21:48 WARNING[1078]
chan_iax2.c: Resyncing the jb. last_delay -150, this delay -2354,
threshold 1300, new offset -181 Feb 17 10:21:48 WARNING[1078]
chan_iax2.c: Resyncing the jb. last_delay -120, this delay -2297,
threshold 1240, new offset 48 Feb 17 10:34:28 WARNING[1078]
chan_iax2.c: Resyncing the jb. last_delay 109, this delay 1556,
threshold 1136, new offset -1556 Feb 17 10:34:28 WARNING[1078]
chan_iax2.c: Resyncing the jb. last_delay -30, this delay -1439,
threshold 1000, new offset -117 Feb 17 10:34:32 WARNING[1078]
chan_iax2.c: Resyncing the jb. last_delay -7, this delay 1608,
threshold 1048, new offset -1725 Feb 17 10:34:32 WARNING[1078]
chan_iax2.c: Resyncing the jb. last_delay -29, this delay -1616,
threshold 1058, new offset -109 Feb 17 10:45:08 WARNING[1078]
chan_iax2.c: Resyncing the jb. last_delay 21, this delay 1751,
threshold 1620, new offset -1751 Feb 17 10:45:08 WARNING[1078]
chan_iax2.c: Resyncing the jb. last_delay -7, this delay 1724,
threshold 1686, new offset -1724 Feb 17 10:45:08 WARNING[1078]
chan_iax2.c: Resyncing the jb. last_delay -60, this delay -1716,
threshold 1000, new offset -8 Feb 17 10:45:08 WARNING[1078]
chan_iax2.c: Resyncing the jb. last_delay -119, this delay -1757,
Resyncing the jb. last_delay 75, this delay 1421, threshold 1326, new
offset -1421 Feb 17 11:28:45 WARNING[1078] chan_iax2.c: Resyncing the
jb. last_delay 274, this delay 1595, threshold 1282, new offset -1595
Feb 17 11:29:03 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay
-1311, this delay 820, threshold 1824, new offset -2415 Feb 17 11:29:03
WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -1349, this
delay 761, threshold 1752, new offset -2182 Feb 17 11:29:03
WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -299, this
delay -2127, threshold 1598, new offset -288 Feb 17 11:29:03
WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay -270, this
delay -2106, threshold 1540, new offset -76 Feb 17 11:46:15
WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 98, this delay
1878, threshold 1206, new offset -1878 Feb 17 11:46:15 WARNING[1078]
chan_iax2.c: Resyncing the jb. last_delay 44, this delay 1799,
threshold 1150, new offset -1799 Feb 17 11:46:15 WARNING[1078]
chan_iax2.c: Resyncing the jb. last_delay 28, this delay 1781,
threshold 1146, new offset -1781 Feb 17 11:46:15 WARNING[1078]
chan_iax2.c: Resyncing the jb. last_delay -150, this delay -1753,
threshold 1000, new offset -46 Feb 17 11:46:15 WARNING[1078]
chan_iax2.c: Resyncing the jb. last_delay -150, this delay -1765,
threshold 1000, new offset -16 Feb 17 11:46:15 WARNING[1078]
chan_iax2.c: Resyncing the jb. last_delay -149, this delay -1747,
threshold 1298, new offset -131 Feb 17 11:54:36 WARNING[1078]
chan_iax2.c: Resyncing the jb. last_delay -44, this delay 1136,
threshold 1064, new offset -1152 Feb 17 11:54:36 WARNING[1078]
chan_iax2.c: Resyncing the jb. last_delay 1, this delay 1155, threshold
Resyncing the jb. last_delay 405, this delay 1547, threshold 1080, new
offset -1547 Feb 17 11:54:36 WARNING[1078] chan_iax2.c: Resyncing the
jb. last_delay -26, this delay 1115, threshold 1054, new offset -1115
Feb 17 11:54:36 WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay
-60, this delay -1133, threshold 1000, new offset -414 Feb 17 11:54:48
WARNING[1078] chan_iax2.c: Resyncing the jb. last_delay 3, this delay
1144, threshold 1048, new offset -1558
The contents of this email message and any attachments are confidential

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Post by Adam Robins
-----------------------------------------------------------------------
-
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Adam Robins

2006-02-23 12:58:41 UTC

Permalink

Thanks,

We already have a cron reboot of all of our Asterisk servers every
night. We've been doing this for over a year due to memory leak issues.

After 2 weeks of messing around with every conceivable IAX2 and
jitterbuffer configuration, I switched to SIP yesterday. Complaints
went from 10-20 per day to ZERO. Literally overnight.

-----Original Message-----
From: asterisk-users-***@lists.digium.com
[mailto:asterisk-users-***@lists.digium.com] On Behalf Of Simone
Cittadini
Sent: Thursday, February 23, 2006 4:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer
Tuning

Post by Adam Robins
Thanks, but we already have the TOS bits set to 0xB8, which matches
the QoS settings in our switches and routers.
This is definitely something that changed in the 1.07 to 1.24 upgrade.
We have a pair of identical 1.07 servers connected via the same
network pipe that do not exhibit these issues.
I might try recompiling with the old jitterbuffer to see if it makes a
difference.
----------------------------------------------------------------------
--

I've not 1.24 in producton yet, still 1.21, anyway I've noticed that
restarting asterisk every night dramatically reduces complaints about
choppy calls (I think is something about a memory leak and not
jitterbuffer, anyway is something easy to do so it's worth trying)
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Martin Joseph

2006-02-23 18:15:12 UTC

Permalink

Post by Zach A
Thanks,
We already have a cron reboot of all of our Asterisk servers every
night. We've been doing this for over a year due to memory leak issues.

??? What do you think this is windows 95??? I had a problem like that I
would be looking at getting rid of asterisk. I don't ;~) I wonder
what your leak is ?

Post by Zach A
After 2 weeks of messing around with every conceivable IAX2 and
jitterbuffer configuration, I switched to SIP yesterday. Complaints
went from 10-20 per day to ZERO. Literally overnight.

I wonder if this is an ILBC frame size issue of some sort? Seems odd.

Adam Robins

2006-02-23 19:34:26 UTC

Permalink

It happened with g729a as well

-----Original Message-----
From: asterisk-users-***@lists.digium.com
[mailto:asterisk-users-***@lists.digium.com] On Behalf Of Martin
Joseph
Sent: Thursday, February 23, 2006 1:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk 1.2.4 IAX2 New Jitterbuffer
Tuning

Post by Zach A
Thanks,
We already have a cron reboot of all of our Asterisk servers every
night. We've been doing this for over a year due to memory leak
issues.

??? What do you think this is windows 95??? I had a problem like that I
would be looking at getting rid of asterisk. I don't ;~) I wonder what
your leak is ?

I wonder if this is an ILBC frame size issue of some sort? Seems odd.

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Noah Miller

2006-02-23 21:45:40 UTC

Permalink

Hi -

Post by Martin Joseph

Post by Adam Robins
After 2 weeks of messing around with every conceivable IAX2 and
jitterbuffer configuration, I switched to SIP yesterday. Complaints
went from 10-20 per day to ZERO. Literally overnight.

I wonder if this is an ILBC frame size issue of some sort? Seems odd.

I've got to add my name to the list here. We're just using GSM over our IAX
links, and our jitterbuffer values look like this:

maxjitterbuffer=1000
resyncthreshold=1000
maxjitterinterps=10

For the most part the new jitterbuffer actually yields much better quality
than the old jitterbuffer, but when the resyncs happen, it's like the call
has a lot of trouble getting get back on track. It flounders for quite a
while, with badly broken audio, sometimes up to 20 seconds before coming
back. I've tried hanging up as soon as event starts happening and then
immediately calling the same number, and the channel comes back with crystal
clarity. So it seems to me like there is something askew with the resync.

- Noah

Rich Adamson

2006-02-23 23:44:16 UTC

Permalink

Post by Noah Miller

Post by Martin Joseph

I wonder if this is an ILBC frame size issue of some sort? Seems odd.

I've got to add my name to the list here. We're just using GSM over our IAX
maxjitterbuffer=1000
resyncthreshold=1000
maxjitterinterps=10
For the most part the new jitterbuffer actually yields much better quality
than the old jitterbuffer, but when the resyncs happen, it's like the call
has a lot of trouble getting get back on track. It flounders for quite a
while, with badly broken audio, sometimes up to 20 seconds before coming
back. I've tried hanging up as soon as event starts happening and then
immediately calling the same number, and the channel comes back with crystal
clarity. So it seems to me like there is something askew with the resync.

If memory serves correctly, I believe I remember Mark applying a fix to
the iax jitterbuffer and that fix had something to do with a counter
rollover or something like that. That fix happened in the last week or
so.

I'm not sure if that would have been included in v1.2.4 or not, but might
be worth a little research.

I also opened a bug a month or two ago involving ilbc and iax, and someone
else confirmed it was a bug. Don't have the bug number handy, but the
problem related to a combination of iax trunking, jitterbuffer and ilbc.
Disabling one of those consistently bypassed the problem.

Adam Robins

2006-02-24 12:53:38 UTC

Permalink

I was using IAX2 with ILBC and no trunking. I also set the
resyncthreshold=-1 to turn it off. Still had major jitter problems.

-----Original Message-----
From: asterisk-users-***@lists.digium.com
[mailto:asterisk-users-***@lists.digium.com] On Behalf Of Rich
Adamson
Sent: Thursday, February 23, 2006 6:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: Asterisk 1.2.4 IAX2 New Jitterbuffer
Tuning

Post by Noah Miller

Post by Martin Joseph

Post by Adam Robins
After 2 weeks of messing around with every conceivable IAX2 and
jitterbuffer configuration, I switched to SIP yesterday.
Complaints went from 10-20 per day to ZERO. Literally overnight.

I wonder if this is an ILBC frame size issue of some sort? Seems

odd.

Post by Noah Miller
I've got to add my name to the list here. We're just using GSM over
maxjitterbuffer=1000
resyncthreshold=1000
maxjitterinterps=10
For the most part the new jitterbuffer actually yields much better
quality than the old jitterbuffer, but when the resyncs happen, it's
like the call has a lot of trouble getting get back on track. It
flounders for quite a while, with badly broken audio, sometimes up to
20 seconds before coming back. I've tried hanging up as soon as event
starts happening and then immediately calling the same number, and the
channel comes back with crystal clarity. So it seems to me like there

is something askew with the resync.

If memory serves correctly, I believe I remember Mark applying a fix to
the iax jitterbuffer and that fix had something to do with a counter
rollover or something like that. That fix happened in the last week or
so.

I'm not sure if that would have been included in v1.2.4 or not, but
might be worth a little research.

I also opened a bug a month or two ago involving ilbc and iax, and
someone else confirmed it was a bug. Don't have the bug number handy,
but the problem related to a combination of iax trunking, jitterbuffer
and ilbc.
Disabling one of those consistently bypassed the problem.

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Pavel Jezek

2006-02-27 16:17:35 UTC

Permalink

I have also issues with jitter over wan (cdma),
I'm trying to debug how dejitter buffer is working (using iax2 jb
debug), but nothing happens/no debug output on asterisk console :-(
is any way how to monitor iax jitter buffer? thx
PJ

Ron Senykoff

2006-02-28 15:49:03 UTC

Permalink

Post by Pavel Jezek
I have also issues with jitter over wan (cdma),
I'm trying to debug how dejitter buffer is working (using iax2 jb
debug), but nothing happens/no debug output on asterisk console :-(
is any way how to monitor iax jitter buffer? thx
PJ

I'm really hoping to see some working settings from some people here.
The jitterbuffer is one of the main features I've been looking forward
to in 1.2. Here are my current settings, if anyone notices a major
problem please let me know. I'm using dropcount of 2 hoping that a
shrink in the jitterbuffer will happen a little faster as a trade-off.
Am I thinking correctly on this? I moved the resyncthreshold way up
since people are having issues with it. My thoughts on
minexcessbuffer=60 is to immediately get a decent buffer going, as
this is much higher than the jitter I usually see (~20ms).

jitterbuffer=yes
dropcount=2
maxjitterbuffer=500
maxexcessbuffer=300
minexcessbuffer=60
jittershrinkrate=1
maxjitterinterps=10
resyncthreshold=1500

Thanks,
-Ron

Pavel Jezek

2006-02-28 16:48:22 UTC

Permalink

Ron, keep in mind, that yoy mix parameters for new and old iax
jitterbuffer implementation, these:
dropcount=2
maxexcessbuffer=300
minexcessbuffer=60
jittershrinkrate=1
maxjitterinterps=10
are ae valid only for _old_ implementation, and I thing, that asterisk
1.2 use new iax buffer by default...

so, I'm using only:
jiterbuffer=yes
forcejitterbuffer=yes
maxjitterbuffer=1500
resyncthreshold=-1
but I don't know, how to monitor if jb is even working, because no
output from iax2 jb debug :-(
can anybody explain?
PJ

Post by Ron Senykoff
I'm really hoping to see some working settings from some people here.
The jitterbuffer is one of the main features I've been looking forward
to in 1.2. Here are my current settings, if anyone notices a major
problem please let me know. I'm using dropcount of 2 hoping that a
shrink in the jitterbuffer will happen a little faster as a trade-off.
Am I thinking correctly on this? I moved the resyncthreshold way up
since people are having issues with it. My thoughts on
minexcessbuffer=60 is to immediately get a decent buffer going, as
this is much higher than the jitter I usually see (~20ms).
jitterbuffer=yes
dropcount=2
maxjitterbuffer=500
maxexcessbuffer=300
minexcessbuffer=60
jittershrinkrate=1
maxjitterinterps=10
resyncthreshold=1500
Thanks,
-Ron
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